Back to EveryPatent.com
United States Patent |
5,297,198
|
Butani
,   et al.
|
March 22, 1994
|
Two-way voice communication methods and apparatus
Abstract
A speakerphone system has a speaker and microphone in a common acoustic
environment and electrically preceded and succeeded, respectively, by
receive and transmit sections and adapted, respectively, to reproduce as
sounds by such speaker signals passed through such receive section from a
remote station, and to convert sounds sensed by such microphone into
signals passed through such transmit section to such station. A computer
derives from the signals from the microphone an electric quantity which is
a measure of the average audible ambient noise level in such environment,
and the computer uses such quantity as one factor among others in
switching signal losses between such two sections so as to change the
state of operation of the system between transmit and receive states. As
an improvement, there is derived from such quantity a control signal which
effects dynamic adjustment of the gain in the receive section so that the
average level of the sounds reproduced by the speaker will increase as
such noise level increases, and conversely (Automatic Level Control).
Other improvements are to provide for the system(a) a plurality of
individually selectable interface circuits for matching the system to
differently designed two-way voice communication channels in the outside
world, (b) an auxiliary microphone for picking up voice sounds in the
environment to be reproduced by the speaker, (c) an additional speaker not
subject to automatic level control, and (d) sound tones produced under
manual control in the environment by the system either to aid an installer
in adjusting the system or to provide a reminder that the system is "on."
Inventors:
|
Butani; Chandru T. (East Hanover, NJ);
Centurrino; Joseph R. (Morris Plains, NJ);
Scheidemann; James F. (Wayne, NJ)
|
Assignee:
|
AT&T Bell Laboratories (Murray Hill, NJ)
|
Appl. No.:
|
815313 |
Filed:
|
December 27, 1991 |
Current U.S. Class: |
379/388.04; 379/388.01; 379/388.02; 379/402; 381/57 |
Intern'l Class: |
H04M 001/00 |
Field of Search: |
379/389,390,402,404,405,419,420
381/57,61
|
References Cited
U.S. Patent Documents
4628526 | Dec., 1986 | Germer | 381/57.
|
4887288 | Dec., 1989 | Erving.
| |
4901346 | Feb., 1990 | Erving.
| |
4959857 | Sep., 1990 | Erving et al.
| |
4965822 | Oct., 1990 | Williams.
| |
5007046 | Apr., 1991 | Erving et al.
| |
5170499 | Dec., 1992 | Grothause | 381/57.
|
Foreign Patent Documents |
0467256 | Jan., 1992 | EP | 381/61.
|
Primary Examiner: Dwyer; James L.
Assistant Examiner: Fournier; Paul A.
Attorney, Agent or Firm: Kip, Jr.; Ruloff F.
Claims
We claim:
1. The improvement in two-way voice telecommunications apparatus comprising
a hands-free speaker and a first hands-free microphone disposed at a local
site in a common acoustic environment, housing means, signal processing
means in said housing means and comprising receive and transmit sections
electrically preceding and succeeding, respectively, said speaker and said
microphone, said receive section having an input and being adapted to
process and then supply to said speaker electrical signals received at
said input and originated at a remote station outside said environment and
representative of voice sounds at said station ("receive signals"), and
said transmit section having an output and being adapted to process and
then supply to said output, for transmission to said station, electrical
signals originated at said microphone and representative of voice sounds
in said environment ("transmit signals"), first and second loss stages in,
respectively, said receive section and said transmit section and
responsive to computer control to produce in said receive and transmit
signals respective signal losses of selected variable value called for by
such control, and a computer coupled to said two stages and responsive to,
among other factors, information from said microphone to dynamically
adjust the losses respectively produced thereby so as to switch the state
of operation of said apparatus between at least transmit and receive
states, and said improvement being that said acoustic environment served
by said speaker is large, said speaker is a loudspeaker adapted to project
into said environment paging announcements and other voice sounds audible
in said environment at multiple locations separated from each other by
more than the normal hearing distance for the normal spoken human voice,
and said improvement also comprising a second auxiliary hands-free
microphone disposed in said environment for converting voice sounds
therein sensed by said electrical auxiliary microphone into electrical
auxiliary voice signals, switch means for selectively effecting a
connection of said electrical auxiliary voice signals to said receive
section for passage therethrough to said loudspeaker, and means responsive
to receive signals incoming to said receive section to disable such
connection such that said electrical auxiliary voice signals do not reach
said loudspeaker and only said receive signals pass through such section
to said loudspeaker.
2. The improvement according to claim 1 in which said computer is
responsive to, among other functions, information from said receive
section to dynamically adjust the loss respectively produced by said two
loss stages, and in which said improvement further comprises means
operable when said auxiliary microphone is connected to said receive
section to render said computer non-responsive to such information when it
is derived from auxiliary voice signals transferred through such receive
section.
3. The improvement in two-way voice telecommunications apparatus comprising
a first hands-free speaker and a hands-free microphone each disposed at a
first local site in a common acoustic environment, housing means, signal
processing means in said housing means and comprising receive and transmit
sections electrically preceding and succeeding, respectively, said speaker
and said microphone, said receive section having an input and being
adapted to process and then supply to said speaker electrical signals
received at said input and originated at a remote station outside
environment and representative of voice sounds at said station ("receive
signals"), and said transmit section having an output and being adapted to
process and then supply to said output, for transmission to said station,
electrical signals originated at said microphone and representative of
voice sounds in said environment ("transmit signals"), first and second
loss stages in, respectively, said receive section and said transmit
section and responsive to computer control to produce in said receive and
transmit signals respective signal losses of selected variable value
called for by such control, and a computer coupled to said two stages to
dynamically adjust the losses respectively produced thereby so as to
switch the state of operation of said apparatus between at least transmit
state and receive states, and said improvement comprising a second
hands-free auxiliary speaker disposed at a second local site acoustically
isolated from said acoustic environment which includes said first local
site, and switch means for selectively effecting a connection of said
auxiliary speaker to said receive section between the input of said
receive section and said loss stage therein such that said auxiliary
speaker is supplied with receive signals incoming to said receive section
without such signals passing through such loss stage to thereby be exposed
to the signal losses produced by such stage.
4. The improvement in two-way voice telecommunications apparatus comprising
a hands-free speaker and a hands-free microphone disposed at a local site
in a common acoustic environment, an amplifier with manually settable gain
preceding such speaker, housing means, signal processing means in said
housing means and comprising receive and transmit sections electrically
preceding and succeeding, respectively, said amplifier and said
microphone, said receive section having an input and being adapted to
process and then supply to said amplifier and speaker electrical signals
received at said input and originated at a remote station outside said
environment and representative of voice sounds at said station ("receive
signals"), and said transmit section having an output and being adapted to
process and then supply to said output, for transmission to said station,
electrical signals originated at said microphone and representative of
voice sounds in said environment ("transmit signals"), first and second
programmable attenuators in, respectively, said receive section and said
transmit section and responsive to computer control to be adjusted to
respective loss settings therefor so as to produce in said receive and
transmit signals respective signal losses of selected variable value
called for by such control, a computer coupled to said two programmable
attenuators and responsive to, among other factors, information from said
microphone to derive from said information two separate indications which
are direct measures of, respectively, the average level of ambient noise
in said environment, and the average level in said environment of voice
sounds present from time to time in said environment, and to dynamically
adjust the respective loss settings of said two programmable attenuators
so as to switch the state of operation of said apparatus between at least
transmit and receive states, and said improvement being that said computer
is responsive to said first indication while being substantially
unresponsive to said second indication to be further operable on only said
first programmable attenuator in said receive section to reset that
attenuator to dynamically adjust the signal loss produced by such
programmable attenuator in receive signals as a function of such noise
level so as to cause voice sounds reproduced by said speaker from such
signals to, over a range, increase in average level when such noise level
increases and decrease in average level when such noise level decreases.
5. The improvement in the method comprising, providing at a local site a
hands-free speaker and microphone in a common acoustic environment for
utilization of said speaker and microphone, respectively, to reproduce as
sound in said environment electrical voice signals received from a remote
station outside said environment ("receive signals") and supplied to said
speaker, and to reproduce voice sounds of people at said site as
electrical voice signals to be transmitted to said station ("transmit
signals"), electrically processing said receive and transmit signals
before being applied to said speaker and transmitted to said station,
respectively, by passing them through an electrical receive section and an
electrical transmit section, respectively, providing a computer, utilizing
said computer to electrically control the processing of said receive and
transmit signals in, respectively, said receive and transmit sections so
as to dynamically adjust losses produced in such sections in such signals
therein and so as to switch the state of operation of the systems
comprising the aforementioned components between at least transmit and
receive states, providing a tone signal generator, and automatically
calibrating said system upon each energization thereof with electricity by
inducing said computer to control said generator in response to said
energization to produce a burst of signals representative of tones of
different frequency, supplying said signal burst to said speaker to be
reproduced thereby as a burst of sound tones projected into said
environment, detecting by said microphone the return lines produced in
sound environment by such sound tone burst, and supplying the electric
signals produced at said microphone from said echos as inputs to said
computer to cause said losses in said sections to be automatically
adjusted through control by said computer so as to calibrate such system
for the acoustic parameters of said environment, and said improvement
comprising, initiating and continuing under manual control of a person a
signal fed to the computer to induce it to control said tone generator to
produce over a time interval determined by that person a succession of
such signal bursts reproduced by said speaker as a corresponding
succession of bursts of sound tones projected into said environment and
usable by an installer of said system to adjust one or more operating
parameters thereof.
6. The improvement in the method comprising, providing at a local site a
hands-free speaker and microphone in a common acoustic environment for
utilization of said speaker and microphone, respectively, to reproduce as
sound in said environment electrical voice signals received from a remote
station outside said environment ("receive signals") and supplied to said
speaker, and to reproduce voice sounds of people at said site as
electrical voice signals to be transmitted to said station ("transmit
signals"), electrically processing said receive and transmit signals
before being applied to said speaker and transmitted to said station,
respectively, by passing them through an electrical receive section and an
electrical transmit section, respectively, providing a computer, utilizing
said computer to electrically control the processing of said receive and
transmit signals in, respectively, said receive and transmit sections so
as to dynamically adjust losses produced in such sections in such signals
therein and so as to switch the state of operation of the systems
comprising the aforementioned components between at least transmit and
receive states, providing a tone signal generator, and automatically
calibrating said system upon each energization thereof with electricity by
inducing said computer to control said generator in response to said
energization to produce a burst of signals representative of tones of
different frequency, supplying said signal burst to said speaker to be
reproduced thereby as a burst of sound tones projected into said
environment, detecting by said microphone the return lines produced in
sound environment by such sound tone burst, and supplying the electric
signals produced at said microphone from said echos as inputs to said
computer to cause said losses in said sections to be automatically
adjusted through control by said computer so as to calibrate such system
for the acoustic parameters of said environment, and said improvement
comprising, initiating and continuing under manual control of a person a
signal fed to the computer to induce it to control said tone generator to
produce a single frequency tone occurring intermittently in successive
bursts of such tone and adapted to remind listeners that such system is in
active operation.
7. The method comprising, providing at a local site a hands-free speaker
and hands-free microphone in a common acoustic environment for utilization
of said speaker and microphone, respectively, to reproduce as sounds in
said environment electrical voice signals received from a remote station
outside said environment ("receive signals") and supplied to said speaker,
and to reproduce voice sounds of people at said site as electrical voice
signals to be transmitted to said station ("transmit signals"),
electrically processing said receive signals before they are applied to
said speaker by passing them through an electrical receive section,
developing via said microphone from audio acoustic energy in said
environment an electrical composite signal comprising first and second
components representative of, respectively, ambient noise in said
environment and voice sounds present from time to time in said environment
electrically analyzing said composite signal to provide a differential
electrical response to said components which discriminates in favor of and
against, respectively, such first and second components, and to derive
from such response a control signal which is a relatively significant
measure of the average level of said noise (the "transmit noise average"),
and is relatively unaffected in value by the presence or absence of such
voice sounds, and dynamically adjusting as a function of the value of said
control signal the gain provided in said receive section for said receive
signals so as, over a range, to increase and decrease the value of such
gain when, respectively, said noise level increases and said noise level
decreases, such dynamic adjusting of said gain superposing upon the gain
of such receive section, as determined by other factors, a dynamic change
in gain which is a function of a corresponding change in such average
ambient noise level, such function being representable by a line graphed
in vertical and horizontal coordinates having scales proportional to
magnitude and representing such gain change and noise level change,
respectively, and such line being in the form of a curve which is
representative of constant intelligibility of said reproduced sounds with
increasing ambient noise level, and the slope of which curve decreases
with increase in noise level.
Description
FIELD OF THE INVENTION
This invention relates generally to methods and apparatus for providing
two-way voice communication between an acoustic environment in which voice
signals are reproduced as sound and a station remote from such environment
and from which such voice signals originate. More particularly, this
invention relates to methods and apparatus of such kind in which the level
of such sound is controlled as a function of the ambient noise in such
environment so as to, over a range, increase when such noise increases,
and conversely.
BACKGROUND OF THE INVENTION
U.S. Pat. No. 5,007,046 issued Apr. 9, 1991 in the name of Richard H.
Erving et al for Computer Controlled Adaptive Speakerphone and assigned to
the assignee hereof and incorporated herein by reference and made a part
hereof (the "'046 patent") discloses a speakerphone adapted to be in an
idle state, a transmit state or a receive state, and to switch from any
one of such states to either of the other two states. The same
speakerphone is also disclosed in U.S. Pat. No. 4,959,857, U.S. Pat. Nos.
4,901,346 and 4,887,288, each assigned to the assignee hereof, but it will
be discussed herein with reference to its disclosure in the '046 patent.
The speakerphone disclosed in that patent, under the control of, for
example, a computer, measures the energy of the incoming transmit and
receive signals and also develops information about the signal and noise
levels for self calibration and efficient operation. This information is
obtained for the computer illustratively by pre-processing analog
circuitry and an analog-to-digital converter. The analog circuitry
converts the incoming transmit and receive signals into a signal that
tracks the envelope of the audio. This envelope information is then
amplified by a logarithmic amplifier which greatly expands the dynamic
operating range of the speakerphone. The resulting analog signals are
passed to the analog-to-digital converter which periodically presents the
computer with digital information corresponding to the logarithm of the
amplitude of the envelope of the signals.
This digital information is used by the computer to develop several
different audio signal averages. A transmit signal average and a receive
signal average are developed by averaging samples of these signals in a
manner that recognizes peaks in the applied signals. Since speech tends to
have many peaks rather than a constant level, this averaging technique
favors detecting speech.
A transmit noise average and a receive noise average are also developed.
The transmit noise average is representative of the noise level of the
operating environment for the speakerphone. The receive noise average
measures the noise level on the line from the far end party. The transmit
noise average and the receive noise average are both developed by
measuring the lowest level seen by the analog-to-digital converter. Since
background noise is generally constant, the lowest level samples provide a
reasonable estimate of the noise level.
The transmit noise average is used in various ways in the operation of the
speakerphone of the '046 patent (the "'046 speakerphone"). When the
speakerphone is in its idle state, an average of the audio informational
signals picked up by the speakerphone microphone from its environment (the
"transmit signal average") is compared to such noise average, and the
exceeding by such signal average of such noise average by more than a
certain threshold, is a factor which, among others, tends to induce
switching from the idle to the transmit state. Similar exceeding by the
transmit signal average of the transmit noise average by more than a
threshold when the speakerphone is in, respectively, the transmit state
and the receive state are factors which, among others, tend, respectively,
to induce retaining of the speakerphone in the transmit state, and the
switching of the speakerphone from the receive state to the transmit
state.
In all of the above cases, however, the transmit noise average has no first
order effect on the level of the voice sounds reproduced by the speaker
element of the speakerphone. Consequently, with progressive increase in
the average ambient noise level of the acoustic environment of that
speaker element, the reproduced voice signals tend to become less and less
intelligible.
SUMMARY OF THE INVENTION
The foregoing disadvantage is overcome according to the present invention
in one of its aspects by the method comprising, providing at a local site
a handsfree speaker and handsfree microphone in a common acoustic
environment for utilization of said speaker and microphone, respectively,
to reproduce as sound in said environment electrical voice signals
received from a remote station outside said environment ("receive
signals") and applied to said speaker, and to reproduce voice sounds of
people at said site as electrical voice signals to be transmitted to said
station, electrically processing said receive signals before their
application to said speaker by passing them through an electrical receive
section, developing via said microphone from audio acoustic energy in said
environment a composite signal comprising first and second components
representative of, respectively, ambient noise in said environment, and
voice sounds present from time to time therein, electrically analyzing
such composite signal to provide, as a differential electrical response to
said components, an electrical quantity which discriminates in favor of
and against, respectively, said first and said second components, and
which is a relatively significant measure of the average level of such
noise while being relatively unaffected in value by the presence or
absence of such voice sounds, deriving from such quantity a control
signal, and dynamically adjusting by said control signal the gain provided
in said receive section for said receive signals so as, over a range, to
increase and decrease the value of such gain when, respectively, said
noise increases and said noise decreases. By the use of such method it is
possible to maintain over a range substantially constant intelligibility
of the voice sounds reproduced in the environment by the speaker as the
level of the ambient noise therein increases while, conversely, it is
possible to reduce such voice sounds to levels which are more pleasing
when such ambient noise level decreases.
BRIEF DESCRIPTION OF THE DRAWING
For a better understanding of the invention, reference is made to the
following description of exemplary embodiments thereof, and to the
accompanying drawings wherein:
FIG. 1 is a block diagram of the two-way speakerphone device which is
disclosed in the '046 patent, and in which is incorporated one or more of
the improvements according to the present invention;
FIG. 2 is a block diagram of part of the left hand end (FIG. 1) of the FIG.
1 speakerphone as modified according to the invention hereof, FIG. 2
showing part of a system according to such invention;
FIG. 3 is a block diagram of part of the right hand end (FIG. 1) of the
FIG. 2 system;
FIG. 4 is a schematic of a programmable attenuator and a low pass filter
employed in a transmit section of the FIG. 1 and FIG. 2 systems;
FIG. 5 is a schematic of a programmable attenuator and a low pass filter
employed in a receive section of the FIG. 1 and FIG. 2 systems;
FIG. 6 depicts a general speakerphone circuit and two types of coupling
that most affect its operation;
FIG. 7 is a state diagram depicting the three possible states of the
speakerphone of FIG. 1;
FIG. 8 depicts a flow chart illustrating the operation of the speakerphone
of FIG. 1 in determining whether to remain in an idle state or move from
the idle state to a transmit or a receive state;
FIG. 9 is a modified state diagram showing a two-state mode of operation
used in the FIG. 2 system and usable in other two-way voice communications
system in various applications of the invention hereof;
FIG. 10 is a flow chart pertaining to controlling in the FIG. 2 system the
level of sound of received signals reproduced by a handsfree speaker in an
acoustic environment as a function of the average ambient noise level
determined as existing in such environment;
FIG. 11 is a diagram graphing the gain obtainable in the electrical receive
section in which various gain setting points may be automatically selected
as a function of such average ambient noise level, the graphed lines in
such diagram relating the level of the voice sounds reproduced in such
environment to such noise level;
FIG. 12 is a schematic diagram in the time-amplitude domain of successive
tone bursts which may be used, according to the invention, by an installer
as an aid in adjusting systems according to the invention; and
FIG. 13 is a schematic diagram in the time-amplitude domain of bursts of a
single frequency tone which may be used according to the invention as a
reminder that the system with which the tone is used is "on".
In the disclosure which follows, the term "gain" is used in the opposite
sense to "loss". That is, a positive gain is a negative loss, and
conversely.
DETAILED DESCRIPTION
FIG. 1 is a functional block representation of the computer controlled
prior art adaptive speakerphone 100 which is disclosed in the '046 patent.
As shown, the speakerphone generally comprises a transmit section 200, a
receive section 300, and a computer 110. A microcomputer commercially
available from Intel Corporation as Part No. 8051 may be used for computer
110 with the proper programming. A microphone 111 couples audio signals to
the speakerphone and a speaker 112 receives output audio signals from the
speakerphone.
By way of operation through illustration, an audio signal provided by a
person speaking into the microphone 111 is coupled into the transmit
section 200 to a multiplexer 210. In addition to being able to select the
microphone speech signal as an input, the multiplexer 210 may also select
calibration tones as its input. These calibration tones are provided by a
calibration circuit 113 and are used, in this instance, for calibration of
the hardware circuitry in the transmit section 200. Circuit 113 is also
referred to herein as a signal generator circuit (FIG. 2) since, according
to the invention hereof, circuit 113 is used for purposes in addition to
calibration.
Connected to the multiplexer 210 is a mute control 211 which mutes the
transmit path in response to a control signal from the computer 110. A
high pass filter 212 connects to the mute control 211 to remove the room
and low frequency background noise in the speech signal. The output of the
high pass filter 212 is coupled both to a programmable attenuator 213 and
to an envelope detector 214. In response to a control signal from the
computer 110, the programmable attenuator 213 inserts loss in the speech
signal in three and one half dB steps up to a total of sixteen steps,
providing 56 dB of total loss. This signal from the programmable
attenuator 213 is coupled to a low pass filter 215 which removes any
spikes that might have been generated by the switching occurring in the
attenuator 213. This filter also provides additional signal shaping to the
signal before the signal is transmitted by the speakerphone over audio
line 101 to a hybrid 610 (FIG. 6). After passing through the envelope
detector 214, the speech signal from the filter 212 is coupled to a
logarithmic amplifier 216, which expands the dynamic range of the
speakerphone to approximately 60 dB for following the envelope of the
speech signal.
The receive section 300 contains speech processing circuitry that is
functionally the same as that found in the transmit section 200. A speech
signal received over an input audio line 102 from the hybrid 610 is
coupled into the receive section 300 to the multiplexer 310. Like the
multiplexer 210, the multiplexer 310 may also select calibration tones for
its input, which are provided by the calibration circuit 113. Connected to
the multiplexer 310 is a mute control 311 which mutes the receive path in
response to a control signal from the computer 110. A high pass filter 312
is connected to the mute control 311 to remove the low frequency
background noise from the speech signal.
The output of the high pass filter 312 is coupled both to an envelope
detector 314 and to a programmable attenuator 313. The envelope detector
314 obtains the signal envelope for the speech signal which is then
coupled to a logarithmic amplifier 316. This amplifier expands the dynamic
range of the speakerphone to approximately 60 dB for following the
envelope of the receive speech signal. The programmable attenuator 313,
responsive to a control signal from the computer 110, inserts loss in the
speech signal in three and one half dB steps in sixteen steps, for 56 dB
of loss. This signal from the programmable attenuator 313 is coupled to a
low pass filter 315 which removes any spikes that might have been
generated by the switching occurring in the attenuator 313. This filter
also provides additional signal shaping to the signal before the signal is
coupled to the loudspeaker 112 via an amplifier 114.
The signals from both the logarithmic amplifier 216 and the logarithmic
amplifier 316 are multiplexed into an eight-bit analog-to-digital
converter 115 by a multiplexer 117. The converter 115 presents the
computer 110 with digital information about the signal levels every 750
microseconds.
The computer 110 measures the energy of the incoming signals and develops
information about the signal and noise levels. Both a transmit signal
average and a receive signal average are developed by averaging samples of
each signal according to the following equation:
##EQU1##
where Sampling rate=1333 per second
s.sub.t =new sample
y.sub.t-1 =old average
y.sub.t =new average
This averaging technique tends to pick out peaks in the signal applied.
Since speech tends to have many peaks rather than a constant level, this
average favors detecting speech.
Both a transmit noise average and a receive noise average are also
developed. The transmit noise average determines the noise level of the
operating environment of the speakerphone. The receive noise average
measures the noise level on the line from the far-end party. The transmit
noise average and the receive noise average are both developed by
measuring the lowest level seen by the converter 115. Since background
noise is generally constant, the lowest samples provide a reasonable
estimate of the noise level. The transmit and receive noise averages are
developed using the following equation:
##EQU2##
where Sampling rate=1333 per second
s.sub.t =new sample
y.sub.t-1 =old average
y.sub.t =new average
This equation strongly favors minimum values of the envelope of the applied
signal, yet still provides a path for the resulting average to rise when
faced with a noisier environment.
Two other signal levels are developed to keep track of the loop gain, which
affects the switching response and singing margin of the speakerphone.
These signal levels are the speech level that is present after being
attenuated by the transmit attentuator 213 and the speech level that is
present after being attentuated by the receive attenuator 313. In the
speakerphone, these two levels are inherently known due to the fact that
the computer 110 directly controls the loss in the attenuators 213 and 313
in discrete amounts, 3.5 dB steps with a maximum loss of 56 dB in each
attenuator. All of these levels are developed to provide the computer 110
with accurate and updated information about what the current state of the
speakerphone should be.
As in all speakerphones, the adaptive speakerphone needs to use thresholds
to determine its state. Unlike its analog predecessors, however, those
thresholds need not be constant. The computer 110 has the ability to
recalibrate itself to counteract variation and aging of hardware circuitry
in the speakerphone. This is achieved by passing a first and a second
computer-generated test tone through the transmit path and the receive
path of the hardware circuitry and measuring both responses.
These test tones are generated at a zero dB level and a minus 20 dB level.
The difference measured between the zero dB level tone and the minus 20 dB
level tone that passes through the speakerphone circuitry is used as a
base line for setting up the thresholds in the speakerphone. First, by way
of example, the zero dB level tone is applied to the transmit path via
multiplexer 210 and that response measured by the computer 110. Then the
minus 20 dB tone is similarly applied to the transmit path via multiplexer
210 and its response measured by the computer. The difference between the
two responses is used by the computer as a basic constant of
proportionally that represents "20 dB" of difference in the transmit path
circuitry, This same measurement is similarly performed on the receive
path circuitry by applying the two test tones via multiplexer 310 to the
receive path. Thus, a constant of proportionality is also obtained for
this path. The number measured for the receive path may be different from
the number measured by the transmit path due to hardware component
variations. The computer simply stores the respective number for the
appropriate path with an assigned value of minus 20 dB to each number.
Once the computer has determined the number representing minus 20 dB for
each path, it is then able to set the required dB threshold levels in each
path that are proportionally scaled to that path's number. Also, because
of the relative scaling, the common thresholds that are set up in each
path always will be essentially equal even though the values of
corresponding circuit components in the paths may differ considerably.
As part of the calibration process, the speakerphone also measures the
acoustics of the room in which it operates. Through use of the calibration
circuit 113, the speakerphone generates a series of eight millisecond tone
bursts throughout the audible frequency of interest and uses these in
determining the time-domain acoustic response of the room. Each tone burst
is sent from the calibration circuit 113 through the receive section 300
and out the leadspeaker 112. The integrated response, which is reflective
of the echoes in the room from each tone burst, is picked up by the
microphone 111 and coupled via the transmit section 200 to the computer
110 where it is stored as a composite response pattern. A more detailed
description of such pattern appears in the '046 patent. This response is
characterized by two important factors: the maximum amplitude of the
returned signal, and the duration of the echoes. The amplitude of the
returned signal determines what level of transmit speech will be required
to break in on receive speech. The greater the acoustic return, the higher
that threshold must be to protect against self-switching. The duration of
the echoes determine how quickly speech energy injected into the room will
dissipate, which controls how fast the speakerphone can switch from a
receive to a transmit state. If the room acoustics are harsh, therefore,
the speakerphone adapts by keeping switching response on a par with that
of a typical analog device. But when acoustics are favorable, it speeds up
the switching time and lowers break in thresholds to provide a noticeable
improvement in performance.
The concept of self-calibration is also applied to the speakerphone's
interface to a hybrid. During a conversation, the computer measures the
degree of hybrid reflection that it sees. This hybrid reflection provides
a measure of both the hybrid and far-end acoustic return. Its average
value is determined using the following equation:
##EQU3##
where Sampling rate=1333 per second
R.sub.t =receive signal average
T.sub.t =transmit signal average
H.sub.t-1 =old hybrid average
H=new hybrid average
This equation develops the hybrid average value by subtracting a transmit
signal from a receive signal and then averaging these signals in a manner
that favors the maximum difference between them. The receive signal is
that signal provided to the speakerphone by the hybrid on the receive line
and the transmit signal is that signal provided to the hybrid by the
speakerphone on the transmit line. By developing an estimate of the hybrid
average, the amount of switched loss required in the speakerphone to
maintain stability may be raised or lowered. By lowering the amount of
switched loss, speakerphone switching operation becomes more transparent
and can even approach full-duplex for fully digital connections.
The estimate of the hybrid average is also used to determine the switching
threshold level of the speakerphone in switching from the transmit state
to the receive state (receive break in). Since the estimate of the hybrid
average is used to develop an expected level of receive speech due to
reflection, additional receive speech due to the far-end talker may be
accurately determined and the state of the speakerphone switched
accordingly.
To obtain an accurate representation of the line conditions, hybrid
averaging is performed only while the speakerphone is in the transmit
state. This insures that receive speech on the receive line during a quiet
transmit interval cannot be mistaken for a high level of hybrid return.
This averaging therefore prevents receive speech, that is not great enough
to cause the speakerphone to go into the receive state, from distorting
the estimated hybrid average.
Another boundary condition employed in developing this hybrid average is a
limitation on the acceptable rate of change of transmit speech. If
transmit speech ramps up quickly, then the possibility of sampling errors
increases. To avoid this potential source of errors, the hybrid average is
only developed during relatively flat intervals of transmit speech (the
exact slope is implementation-dependent).
To ensure stable operation with an adaptive speakerphone in use at both the
near-end and the far-end by both parties, the amount that the hybrid
average may improve during any given transmit interval is also limited. In
the adaptive speakerphone 100, for example, the hybrid average is allowed
to improve no more than 5 dB during each transmit state. In order for the
hybrid average to improve further, a transition to receive and then back
to transmit must be made. This insures that the far-end speakerphone has
also had an opportunity to go into the transmit state and has similarly
adapted. Thus, each speakerphone is able to reduce its inserted loss down
to a point of balance in a monotonic fashion. Limiting the amount of
change in the hybrid average during a transmit interval also allows this
speakerphone to be operable with other adaptive speakerphones such as
echo-cancelling speakerphones that present a varying amount of far-end
echo as they adapt.
For ease of operation and for configuring the speakerphone, a user
interface 120 through which the user has control over speakerphone
functions is provided internal to the speakerphone 100. Such user
interface is comprised in part in the FIG. 2 system of a control panel 40
(FIG. 3).
Referring now to FIG. 4, there is shown a detail schematic of the
programmable attenuator 213. This attenuator comprises multiple sections
which are formed by passing the output of an amplifier in one section
through a switchable voltage divider and then into the input of another
amplifier. The signal on line 142 from the high pass filter 212 is coupled
directly to a first section of the attenuator 213 comprising a voltage
divider consisting of resistors 222 and 223, a switch 224 and a follower
amplifier 226. When the switch 224 is closed shorting resistor 222, the
voltage developed across the voltage divider essentially will be the
original input voltage, all of which develops across resistor 223. Once
the switch is opened, in response to a command from the computer 110, the
signal developed at the juncture of resistors 222 and 223 is reduced from
that of the original input voltage level to the desired lower level. The
loss is inserted in each section of the attenuator in this manner.
Thus in operation, a speech signal passing through the first section of the
attenuator is either passed at the original voltage level or attenuated by
28 dB. If the switch is turned on, i.e., the resistor 222 shorted out,
then no loss is inserted. If the switch is turned off, then 28 dB of loss
is inserted. The signal then goes through a second similar section which
has 14 dB of loss. This second section of the attenuator 213 comprises a
voltage divider consisting of resistors 227 and 228, a switch 229 and a
follower amplifier 230. This second section is followed by a third section
which has 7 dB of loss. This third section of the attenuator 213 comprises
a voltage divider consisting of resistors 231 and 232, a switch 233 and a
follower amplifier 234. A fourth and final section has 31/2 dB of loss.
This final section of the attenuator 213 comprises resistors 235 and 236
and a switch 237. By selecting the proper combination of on/off values for
switches 224, 229, 233 and 237, the computer 110 may select from 0 to 56
dB of loss in 31/2 dB increments. It should be understood that if a finer
control of this attenuator is desired such that it could select
attenuation in 1.75 dB increments, it is but a simple matter for one
skilled in the art, in view of the above teachings, to add another section
to the attenuator thereby providing this level of control.
This signal from the programmable attenuator 213 is coupled to the low pass
filter 215 which provides additional shaping to the transmit signal. Low
pass filter 215 comprises a follower amplifier 238, and associated
circuitry comprising capacitors 239 and 240, and resistors 241 and 242.
The output of filter 215 is coupled to a transmit audio output level
conversion circuit, comprising amplifier 144, resistors 145, 146 and 147,
and also capacitor 148, for connection to the audio output line 101. This
output level conversion circuit provides an output impedance of 600 ohms
for matching to the output line 101.
Referring now to FIG. 5, there is shown a detail schematic for the
programmable attenuator 313, the low pass filter 315 and the amplifier 114
for the loudspeaker 112. The same basic components are used in
implementing the programmable attenuator 313 and the programmable
attenuator 213. Because of this and the detailed description given to
attenuator 213, this attenuator 313 will not be described in similar
detail.
Follower amplifiers 326, 330 and 334 along with resistors 322, 323, 327,
328, 331, 332, 335 and 336, and also switches 324, 329, 333 and 337
combine in forming the four sections of the attenuator 313. As in
attenuator 213, a speech signal is attenuated 28 dB by section one, 14 dB
by section two and 7 dB and 31/2 dB by sections three and four
respectively.
The signal from the programmable attenuator 313 is coupled to the low pass
filter 315 which provides additional shaping to the receive signal. Low
pass filter 315 comprises a follower amplifier 338, and associated
circuitry including capacitors 339 and 340, and resistors 341 and 342. In
amplifier 114, an amplifier unit 149 and associated circuitry, variable
resistor 150, resistors 151 and 152, and capacitors 153 and 154, provide
gain for the output signal from low pass filter 315 before coupling this
signal to the speaker 112 via a capacitor 155.
Further details of the FIG. 1 speakerphone are disclosed in the '046
patent.
With reference to FIG. 6, there is shown a general speakerphone circuit 600
for describing the two types of coupling, hybrid and acoustic, that most
affect the operation of a speakerphone being employed in a telephone
connection. A hybrid 610 connects the transmit and receive paths of the
speakerphone to a telephone line whose impedance may vary depending upon,
for example, its length from a central office, as well as, for example,
other hybrids in the connection. And the hybrid only provides a best case
approximation to a perfect impedance match to this line. Thus a part of
the signal on the transmit path to the hybrid returns over the receive
path as hybrid coupling. With this limitation and the inevitable acoustic
coupling between a loudspeaker 611 and a microphone 612, transmit and
receive loss controls 613 and 614 are inserted in the appropriate paths to
avoid degenerative feedback or singing.
In accordance with the invention, the computer controlled adaptive
speakerphone 100 of FIG. 1 advantageously employs a process or program
described herein with reference to a state diagram of FIG. 7 and the flow
diagram of FIG. 8 for improved performance. This process dynamically
adjusts the operational parameters of the speakerphone for the best
possible performance in view of existing hybrid and acoustic coupling
conditions.
Referring now to FIG. 7, there is shown the state diagram depicting the
possible states of the speakerphone 100. The speakerphone initializes in
an idle state 701. While in this state, the speakerphone has a symmetrical
path for entering into either a transmit state 702 or a receive state 703,
according to which of these two has the stronger signal. If there is no
transmit or receive speech while the speakerphone is in the idle state
701, the speakerphone remains in this state as indicated by a loop out of
and back into this idle state. Generally, if speech is detected in the
transmit or receive path, the speakerphone moves to the corresponding
transmit or receive state. If the speakerphone has moved to the transmit
state 702, for example, and transmit speech continues to be detected, the
speakerphone then remains in this state. If the speakerphone detects
receive speech having a stronger signal than the transmit speech, a
receive break-in occurs and the speakerphone moves to the receive state
703. If transmit speech ceases and no receive speech is present, the
speakerphone returns to the idle state 701. Operation of the speakerphone
in the receive state 703 is essentially the reverse of its operation in
the transmit state 702. Thus if there is receive speech following the
speakerphone moving to the receive state 703, the speakerphone stays in
this state. If transmit speech successfully interrupts, however, the
speakerphone goes into the transmit state 702. And if there is no receive
speech while the speakerphone is in the receive state 703 and no transmit
speech to interrupt, the speakerphone returns to the idle state.
In the modified speakerphone shown by FIGS. 2-5 and incorporating the
inventive improvements hereof, the speakerphone's operation may have the
three states depicted in FIG. 7. It is preferred, however, that such
improved speakerphone have only transmit and receive states as depicted in
FIG. 9. For such two state mode of operation, the transmit state 901 is
the preferred state in which the speakerphone initializes and in or to
which it will stay or return in the absence of both transmit and receive
signals. If there is no transmit or receive speech while the speakerphone
is in the state 901, the speaker phone remains in this state as indicated
by a loop 902 out of and back into this state. If the speakerphone is in
the transmit state, for example, and transmit speech occurs and continues
to be detected as represented by loop 903, the speakerphone then remains
in this state. If the speakerphone detects receive speech having a
stronger signal than the transmit speech, a receive break-in occurs and
the speakerphone moves as indicated by shift line 904 to the receive state
905. Operation of the speakerphone in the receive state is essentially the
reverse of its operation in the transmit state. Thus, if there is receive
speech following the speakerphone moving to the receive state as depicted
by loop 906, the speakerphone stays in this state. If transmit speech
successfully interrupts, however, the speakerphone goes, as depicted by
shift line 907, into the transmit state 901. And if there is no receive
speech while the speakerphone is in the receive state 905 and no transmit
speech to interrupt, the speakerphone returns to the transmit state 901.
Referring now to the FIG. 1 speakerphone having three states of operation,
there is shown in FIG. 8 a flow chart illustrating in greater detail the
operation of the speakerphone 100 in determining whether to remain in the
idle state or move from the idle state to the transmit state or receive
state. The process is entered at step 801 wherein the speakerphone is in
the idle state. From this step, the process advances to the decision 802
where it determines whether the detected transmit signal is greater than
the transmit noise by a certain threshold. If the detected transmit signal
is greater than the transmit noise by the desired amount, the process
proceeds to decision 803. At this decision, a determination is made as to
whether the detected transmit signal exceeds the expected transmit signal
by a certain threshold.
The expected transmit signal is that component of the transmit signal that
is due to the receive signal coupling from the loudspeaker to the
microphone. This signal will vary based on the receive speech signal, the
amount of switched loss, and the acoustics of the room as determined
during the acoustic calibration process. The expected transmit level is
used to guard against false switching that can result from room echoes;
therefore, the transmit level must exceed the expected transmit level by a
certain threshold in order for the speakerphone to switch into the
transmit state.
If the detected transmit signal does not exceed the expected transmit
signal by the threshold, the process advances to decision 806. If the
detected transmit signal exceeds the expected transmit signal by the
threshold, however, the process advances to step 804 where a holdover
timer is initialized prior to the speakerphone entering the transmit
state. Once activated, this timer keeps the speakerphone in either the
transmit state or the receive state over a period of time, approximately
1.2 seconds, when there is no speech in the then selected state. This
allows a suitable period for bridging the gap between syllables, words and
phrases that occur in normal speech. From step 804 the process advances to
step 805 where the speakerphone enters the transmit state.
Referring once again to step 802, if the detected transmit signal is not
greater than the transmit noise by a certain threshold, then the process
advances to the decision 806. In this decision, and also in decision 807,
the receive path is examined in the same manner as the transmit path in
decisions 802 and 803. In decision 806, the detected received signal is
examined to determine if it is greater than the receive noise by a certain
threshold. If the detected receive signal is not greater than the receive
noise by this threshold, the process returns to the step 801 and the
speakerphone remains in the idle state. If the detected receive signal is
greater than the receive noise by the desired amount, the process proceeds
to decision 807. At this decision, a determination is made as to whether
the detected receive signal exceeds the expected receive signal by a
certain threshold.
The expected receive signal represents the amount of speech seen on the
receive line that is due to transmit speech coupled through the hybrid.
This signal is calculated on an ongoing basis by the speakerphone and
depends on the hybrid average, the amount of switched loss, and the
transmit speech signal. Since the transmit speech path is open to some
extent while the speakerphone is in the idle state, this causes a certain
amount of hybrid reflection to occur, which, in turn, causes a certain
amount of the speech signal detected on the receive path to be due to
actual background noise or speech in the room. This, in turn, is read as a
certain expected level of receive speech. And the actual receive speech
signal must surpass this expected level by the threshold in order for the
speakerphone to determine with certainty that there is actually a far-end
party talking.
If the detected receive signal does not exceed the expected receive signal
by the threshold, the process returns to the step 801 and the speakerphone
remains in the idle state. If the detected receive signal exceeds the
expected receive signal by the threshold, however, the process advances to
step 808 where the holdover timer is initialized. From step 808 the
process advances to step 809 where the speakerphone is directed to enter
the receive state.
Once the speakerphone has assumed the transmit state as described in
connection with FIG. 8, the processing continues to determine if the
system should remain in the transmit state or switch to the receive state
or the idle state. Similarly, if the system has assumed the receive state
as described in connection with FIG. 8, the processing continues to
determine if the system should remain in that state or switch to the
transmit state or the idle states. In both instances one of the factors
which affects the decisions made in the course of the processing is the
value of the quantity TX-N (see FIG. 8) which is a measure of the ambient
noise which is present in the environment of microphone 111 and is sensed
thereby, and which is calculated by computer 110 as afore described. As
part of such determinations, the average level of any transmit speech
occurring is compared to TX-N, and decisions are made on the basis of
whether or not such speech level exceeds TX-N by a certain threshold. For
a better understanding of the steps occurring and the decisions made in
the course of such processing when the FIG. 1 system is in the transmit
state or the receive state, reference is made to the '046 patent. It
suffices to say here that, if the system is a three state system as
depicted in FIG. 7, the ambient noise level of the system's environment
is, for whatever one of the three states that system is then in, one of
the factors determinative of whether the system over time will remain in
that state or shift to a different state. What has just been said for such
three state system is true also for the two state system depicted in FIG.
9.
Reference is now made to FIG. 2 which shows various improvements
incorporated according to the invention hereof in the front end of the
FIG. 1 speakerphone 100 earlier described. In FIG. 2, a dash line 20
represents the demarcation between that front end and the rest of the
speakerphone. In the figure, the reference numeral 21 designates the
housing (depicted by a dot-dash line) for the speakerphone case 22
containing much of the speakerphone circuitry. The reference numeral 23
designates an acoustic environment (depicted by a dash line) which is
common to the speakerphone microphone 111 and the main speaker of the FIG.
2 system, such microphone and speaker being disposed at a local site
within that environment. The case 22 may be optionally located either
within or without environment 23.
Housing 21 mounts on its outside a jack 25 into which may be plugged any
one of a plurality of external conductor means which are two-way signal
conducting in the sense that they conduct electric signals in both
directions over the same conductor(s). Such external bidirection signal
conducting means or channels are adapted to couple for communication
purposes the speakerphone 100 to a selected remote station disposed
outside environment 23. One such channel is represented by the shown
telephone line 26 which consists of tip and ring leads 27, 28 terminating
in a plug 29 adapted to be selectively fitted into and removed from jack
25, and which line 26 may be coupled to a conventional telephone set.
These various two-way signal conducting channels have different
characteristics, and they may, for example, consist (besides line 26) of a
telephone system analog station port, a telephone system dry loop port and
a telephone system loop start trunk port.
Interposed between jack 25 and the connection 620 of hybrid 610 to the
outside world are a plurality of interface circuits I.sub.a -I.sub.d
(designated as 30-33) having different electrical characteristics and
adapted to match corresponding ones of the mentioned channels to the
hybrid. Any selected one of these interface circuits may be electrically
coupled between jack 25 and the hybrid by a multisection mechanical
selector switch 34. Switch 34 has a first section comprising a movable
contact 35 angularly movable by a linkage 36 to any one of a number of
fixed positions and adapted at each of them to connect to jack 25 one of
circuits 30-33. The same switch has a second section comprising a movable
contact 37 similarly angularly movable by linkage 36 to connect to hybrid
610 the output of that interface circuit of which the input is connected
through contact 35 to jack 25. While the lines connected to the inputs and
outputs of circuits 30-33 are depicted in FIG. 2 as being single leads, it
will be understood that these and other conductor means shown in the
figures may consist of two or more leads so as, in the two lead case, for
example, to have leads corresponding, for example, to the tip and ring
leads of line 26.
The linkage 36 is adapted to move contacts 35 and 37 by angular movement by
hand of a pivoted interface selector switch lever 38 (FIG. 3) mounted on a
control panel 40 disposed on the outside of the housing 21 of the
speakerphone case 22.
As another improvement feature of the FIG. 2 speakerphone, it has, in
addition to its main microphone 111, an auxiliary microphone 50 which
normally is (but need not be) located within the mentioned acoustic
environment 23. The microphones 111 and 50 differ in that voice signals
from the former are, when the speakerphone is connected to a remote
station, passed through transmit section 200 and hybrid 610 to an outside
communications channel to be reproduced as voice sounds at such station
(i.e., not by the speakers of the FIG. 2 speakerphone). In contrast, voice
signals from auxiliary microphone 50 are passed through receive section
300 to be reproduced as voice sounds by the speaker(s) of the speakerphone
system. The manner in which such auxiliary voice signals are so reproduced
is as follows.
The line 51 from microphone 50 terminates in a plug 52 selectively fittable
into and removable from a jack 53 mounted on the outside of housing 21 of
speakerphone case 22. Jack 53 is coupled to the "lower" input 54 of a
multiplexer 55 having an "upper" input 56. Circuit 55 is biased to
normally pass signals on its lower input. The lead 102 from hybrid 610 is
connected to such upper input. The output 57 of multiplexer 55 is
connected to the receive signal input of the multiplexer 310 in receive
section 300. The FIG. 2 speakerphone thus differs from that of FIG. 1 in
that the multiplexer 55 with its normally signal blocking input 56 is
interposed in the path for receive signals between hybrid 610 and the
receive section.
The selection of whether multiplexer 55 passes signal from its input 54 or
its input 56 is controlled by a circuit 58 for detecting signals on a
control lead 59 connected to all of the interface circuits 30-33. The
operation is as follows. In the absence of there being a telephone call in
progress between the speakerphone system and a remote station, there will
be no signal in lead 59 and, since multiplexer 55 is biased to pass
signals on its lower input, any voice signals produced by the microphone
50 will be passed through multiplexer 55 to the receive section 300 to be
reproduced as sound by the speakerphone speaker(s). Assume, however, that
a telephone call is initiated between the speakerphone and a remote
station. For the duration of such call, whichever of interface circuits is
then connected between jack 25 and the hybrid 610 will develop on lead 59
a control signal which is detected by detector circuit 58. That circuit
responds to such signal to control multiplexer 55 to reject signals on its
input 54 and pass signals on its input 56. Thus, so long as such telephone
call is in progress the auxiliary voice signals from microphone 50 are
locked out from reaching the receive section 300 which, hence, then
processes exclusively the receive signals from the remote station. When,
however, that call ceases, the control signal on lead 59 disappears and
the multiplexer 55 reverts to passing any voice signals from microphone 50
through the receive section and to the one or more speakerphone speakers.
In reaching the lower one of such speakers (FIG. 2), those signals must
first pass through the programmable attenuator 313 so as to undergo
whatever signal loss (or gain) as is then set into that circuit. As later
described in detail, the fact that the signals to the lower speaker must
pass through attenuator 313 permits the sounds reproduced by that speaker
from those auxiliary signals to be characterized by automatic level
control ("ALC").
The control signal generated on lead 59 on initiation of a telephone call
is supplied not only to detector circuit 58 but also, via branch lead 65
to computer 110 to switch it to an off-hook condition. In that condition,
the computer is fully active and performs its calculations and control
functions which have been earlier described. Such calculations include the
computation of the mentioned quantity TX-N representative of the transmit
noise level, i.e., the average noise level sensed by microphone 111 in
environment 23, and such control functions include dynamically adjusting
the attenuators 213, 313 to change the losses (or gains) imparted thereby
to the voice signals passing therethrough.
The cessation of the control signal on termination of the telephone call
switches computer 110 back to an on-hook condition. In that latter
condition, the computer remains energized but is inactive in the sense
that it no longer performs such calculations and performs such functions.
In the on-hook condition, however, the computer continues to maintain the
attenuators 213, 313 at the loss (gain) settings last attained thereby in
the course of the immediately preceding telephone call.
In some applications it may be desirable for the FIG. 2 speakerphone to
include an option which prevents the onset of telephone calls from
terminating the connection of the auxiliary microphone to the one or more
speakers. One way for providing such option is for the movable contacts 35
and 37 of the switch 34 to be movable to and from an extra "auxiliary
preempt" position (not shown) which is to the left (FIG. 2) of the shown
fixed contacts contactable by such movable contacts, and at which extra
position the movable ends of these movable contacts do not make electrical
contact with anything or, alternatively, are grounded. When such movable
contacts are at that extra position, no telephone calls or associated
signals can pass through the interface circuits 30-33. Accordingly, those
circuits cannot generate on lead 59 the control signal which locks out
from passing through the multiplexer 55 the auxiliary voice signals from
microphone 50.
Considering now the matter of sound reproduction, in the FIG. 1
speakerphone its speaker 112 is normally (but need not be) disposed within
the speakerphone case on the inside of an apertured grillwork formed in
the speakerphone housing. Also, the amplifier 114 for the speaker 112 has
fixed gain. The FIG. 2 speakerphone system is, however, in one of its
applications, designed for use as a two-way paging system in a large size
acoustic environment as, say, a factory space. For that application, the
FIG. 2 speakerphone has a speaker 70 and an amplifier 71 for driving that
speaker. The level of the sounds reproduced by speaker 70 is subject to
automatic level control (as later described), and speaker 70 is a
loudspeaker in the sense that such level at which voice signals are
reproduced as sounds thereby is high enough so that such sounds are
normally audible at multiple locations within environment 23 which are
spaced from each other by greater than the distance over which normal
speech between persons is audible. Because of such high level of sound
from speaker 70, the distance between such speaker and the microphone 111
(and, also the microphone 50 if used in environment 23) is increased as
compared to the microphone-speaker distance in FIG. 1 in order to avoid
excessive gain around the loop depicted in FIG. 6. Alternatively, such
excessive gain may be avoided, if there is only one microphone, by having
that microphone take the form of two microphone elements which are mounted
back-to-back on speaker 70 and have cardioid sound reception patterns
facing away from each other, and which microphone elements are
electrically connected in opposing relation so that there will be a
balancing out of voice signals produced by the two microphone elements in
response to the same sounds from speaker 70.
The amplifier 71 which drives loudspeaker 70 is a variable gain amplifier
normally contained within case 22 and having a gain control knob (or other
instrumentality) 72 mounted either on the amplifier housing or on the
control panel 40 (FIG. 3) of the speakerphone system. The gain of
amplifier 72 is adapted to be statically set by either an installer or a
user of the FIG. 2 system.
In addition to the speaker 70 subject to ALC, the FIG. 2 system may
include, as shown, an additional loudspeaker 75 located within its own
acoustic environment 76. Environment 76 is acoustically isolated from the
acoustic environment 23 of microphone 111 and loudspeaker 70. The
environment 76 does not, therefore, have present therein the ambient noise
of environment 23. The environment 76 may, but need not be, for example a
room enclosed by sound proof walls which shields the room's interior from
outside noise.
Because environment 76 is not subject to the ambient noise of environment
23 there would be no point in having the level of sounds from speaker 75
increase as such ambient noise increases. Therefore, to avoid having those
sounds be subject to the automatic level control (ALC) to which the sounds
from speaker 70 are subject, the voice signals which feed speaker 75 are
tapped from receive section 300 at a point 77 which precedes attenuator
313. From that point, such signals are supplied by a line 78 to a high
gain driver amplifier 79 for loudspeaker 75 and from that amplifier to the
loudspeaker itself. Amplifier 79 normally is contained within case 22. The
amplifier has a gain control knob (or other instrumentality) 80 which may
be mounted on the amplifier 70 or on control panel 40 and which may be
adjusted to various static gain settings for the amplifier by an installer
or user of the FIG. 2 system.
Turning now to FIG. 3 which shows aspects of other improvements
incorporated in the FIG. 2 system, the control panel 40 has mounted
therein a MUTE switch (also present in the FIG. 1 system) operated by a
slide button shiftable between a left position and a right position at
which the switch actuates computer 110 to cause MUTE circuits 211, 311 to
permit and prevent, respectively, the conduction of signals therethrough.
Another ON/OFF switch has on panel 40 a manually depressible button 81
which is spring loaded to normally be in outward position. Such switch is
employed by the user of the FIG. 2 system to make outgoing telephone
calls. A first depression of the button 81 initiates such a call which
becomes self-held when the button is thereafter released. A second
depression of the button 81 terminates that call. Outgoing calls so
initiated by manipulation of button 81 have the same effect with respect
to the described creation and cessation of a control signal on lead 59
(FIG. 2) as do telephone calls initially incoming to the FIG. 2 system.
Further included on the control panel 40 is a VU meter 82 provided by an
array of similar rectangular vertically elongated light emitting diodes 83
("LED's") which are spaced in side-to-side relation along, and vertically
bisected by, a common horizontal axis. Diodes 83 are electrically
connected to respond to signals applied to the meter 82 to collectively
produce luminous bar displays of variable lengths (from the array's left
hand end) which represent in dB or absolute value the respective
magnitudes of such signals. Signals representing various quantities may be
so applied to meter 82 by a meter selector switch 84 comprising a movable
contact 85 electrically fixedly connected to meter 82 and positionable by
movement by hand of a selector lever 86 on panel 40 to selectively contact
any one of a plurality of fixed contacts electrically coupled to computer
110 and represented in FIG. 3 by contacts 87 and 88. These fixed contacts
87, 88 may be, by way of example, without restriction, have supplied
thereto signals from computer 110 representative of the average level of,
respectively, receive signals reproduced as sound by the FIG. 2 system,
and the ambient noise in environment 23 in which such signals are
reproduced as sound by loudspeaker 70. The meter 82 will then respond to
shifting of selector lever 86 to provide luminous bar displays of either
such average signal level or such average noise level.
A description has earlier been given of how the FIG. 1 speakerphone system
automatically calibrates itself to take into account the acoustics of the
environment in which that system operates. That automatic self-calibration
takes place each time the system is energized (as, for example, when it is
first installed or is subsequently re-energized after being de-energized
for some reason). During each such calibration, the system obtains
information on the acoustics of its environment by generating by its
calibrator circuit 113 and automatically emitting by its speaker 112 a
single train of sound tones of which each tone is of single frequency and
lasts eight milliseconds, but which individual tones in the train are of
progressively increasing frequency over the duration of the entire train
of tones. The acoustic response of such environment to such single train
of tones is detected by microphone 111 and converted by it into signals
which are fed to computer 110 to become a factor affecting the
calculations and control functions performed thereby. In the FIG. 1
system, however, the production of the single train of tones is entirely
automatic and is not under the control of the user of the system.
In the FIG. 2 system in contrast, the system not only automatically
produces such single train of sound tones as described above but, also, is
capable, at the option of the system user, of being actuated to produce
under manual control successive reiterations of such train so as to
generate a sequence of sound tone trains over a time period lasting as
long as desired by the user.
FIG. 12 depicts such a sequence in which a succession of identical trains
90 of sound tones appear in time separated relation, and in which each
train 90 consists of a group of sound tones which are each of a single
frequency but which, among each other, increase in frequency with increase
in time.
Any such sequence of time separated identical sound tone trains is "called
out" from the FIG. 2 system by the use of a tone generator enable switch
92 adapted to be actuated by a manually depressible button 93 mounted on
panel 40 and spring biased to be normally in outward position. When button
93 is first depressed, the switch 92 is actuated to signal computer 110 to
control signal generator 113 to produce tone signals which are reproduced
by the speaker 70 as successive intermittent reiterations of the mentioned
single sound tone train. The computer 110 will continue to cause such
reiterations until button 93 is depressed again to indicate to the
computer to thereby end the period of generation of such sound tone
trains. An effect of the continuance of such period is to disable the
computer 110 from responding to the trains of sound tones so generated.
Thus, it is possible by manipulation of the switch button 93 for the user
of the FIG. 2 system to cause its speaker 70 to project into environment
23 a succession of tone separated sound tone trains of the kind described
over a tone period selected by that user. While, as stated, the response
of such environment to these manually induced sound tone trains is not
utilized to affect the operation of computer 110, these trains nonetheless
have a useful purpose in that the sounds in these trains are clearly
audible to an installer and can be used by that person, when installing or
partly relocating the system, as an aid in making adjustment thereto as,
say, the setting of the static gain provided by amplifier 71.
The FIG. 2 speakerphone system is also adapted at the option of the user to
generate a reminder or warning signal which is reproduced as sound by
speaker 70, and which reminds persons in environment 23 that the system is
"on" and is capable of transmitting what they say to another party who has
activated the system by placing a telephone call thereto.
The possibility of automatically activating the FIG. 2 system when
unattended by a telephone call from a remote station is considered a
desirable feature of the system because it, say, permits persons in the
same building as the FIG. 2 system to use the unattended system for paging
purposes. If however the system has that feature, then it is desirable
that the system sound the mentioned reminder signal in the instance where
the system is in an "on" state permitting transmission of sound therefrom
to such station.
The mentioned reminder signal is shown in FIG. 13 and consists of a tone
burst 94 reproduced as sounds by speaker 70 and separated from each other
by, say, thirty (30) second time intervals. Each of such tone burst
consists of a tone having a single frequency. The cutting in and out of
such reminder signal is effected by a switch 95 comprising a movable
contact 96 adapted to open and close with a fixed contact 97 by the
shifting by hand of a pivoted switch control lever 98 on panel 40. The
reminder signal is initiated by moving lever 98 down to close contact 96
with contact 97 to make a connection which signals computer 110 to control
signal generator 113 to produce an electrical signal converted by speaker
70 into the mentioned of tone burst. So long as movable contact 96 is
closed with fixed contact 97, those tone bursts will be generated and can
be heard by the casual listener independent of whether or not the FIG. 2
system is attended by a person assigned to operate it. The single tone
bursts can, of course be terminated by moving switch lever 98 up to
produce an open between contacts 96 and 97.
The improvements so far disclosed in detail have been improvements in
apparatus or in apparatus together with method. We turn now to a most
important improvement, according to the invention, which is an improvement
in method and which involves the signal or quantity designated herein as
TX-N and representative of the average level of the audio frequency
ambient noise present in the common acoustic environment 23 of the
microphone 111 and the speaker 70.
Considering in further detail the derivation of TX-N, the acoustic energy
in environment 23 comprises two components in the audio frequency range,
namely, a first component constituting audible ambient noise which is
continuously present, and a second component constituting voice sounds
which are present from time to time in environment 23. Those voice sounds
may be subdivided into (a) the voice sounds directly detected by the
microphone from people located in environment 23, (b) the sounds
reproduced by speaker 70 of the voices of persons at remote stations in
communication with the speakerphone system serving environment 23, and (c)
voice sounds which are initiated by people in that environment and are
originally picked up by microphone 111 and converted by it into electric
signals, but a fraction of which signals "leak" in the FIG. 6 loop through
hybrid 610 to be reproduced by speaker 70 as voice sounds, and to
constitute feedback in such loop. Of course voice sounds and signals which
are feedbacks may make more than one pass around that loop, but the
parameters of that loop are controlled by computer 110 so that the loop
gain will not exceed 1.0 so as to cause singing.
Because microphone 111 senses these first and second components in the
acoustic energy in environment 23, the electric signal derived from
microphone 111 from such sensing is a composite signal also comprising
first and second components representative of, respectively, ambient noise
in said environment and voice sounds present therein and possibly of
multiple origin as set out above.
As earlier described, that composite signal is electrically analyzed to
provide a differential electrical response to its first and second
components. That is, such electrical analysis is conducted by the use of
two equations earlier set out, and of which the first employs an averaging
technique which tends to pick out peaks in the component signals so as to
yield a quantity which is primarily a measure of the second "voice"
component of the composite signal. It is, however, the second of the
equations earlier set out which is of interest in relation to the
improvement now being described.
Examining that second equation closely, it will be evident from it that the
processing of the composite signal in accordance with it is as follows.
First the preliminary steps earlier described are carried out of detecting
the signal envelope of the composite signal and of obtaining, by
analog-to-digital conversion, digital samples of magnitude values of such
signal.
After those preliminary steps, the processing continues by deriving a
running average of the value of ones of such samples up to the next sample
to occur, changing the value of such average upon occurrence of such next
sample to reflect its value in such average by obtaining the difference
between the absolute value of such next sample, and the value of such
running average, adding to such running average value, when such
difference is positive, the value of such difference when multiplied by a
first weighting factor, adding to such running average value, when such
difference is negative, the value of such difference multiplied by a
second weighting factor much greater than such first factor, and obtaining
by the foregoing steps an electrical quantity which is a representation of
said running average value as so continuously updated by such additions
thereto of said differences as they occur. All of such post-preliminary
steps are carried out by computer 110. The quantity just mentioned is the
transmit noise quantity TX-N.
From inspection of the mentioned second equation, it will be evident that
the first weighting factor referred to has the value 1/4096 whereas the
second weighting factor has the value 1/4. That is, the first factor has a
value more than a thousandfold less than the value of the second factor.
An effect of such difference is that, among the rapidly occurring changes
in peak magnitude induced in the signal envelope of the composite signal
almost entirely by its voice component, the value of TX-N will respond
only very slowly to those of such changes which are rises or constitute a
rising trend, but the TX-N value will respond quickly to those of such
changes which are negative or constitute a falling trend in magnitude. A
consequence of this asymmetry of the TX-N response (which asymmetry is due
to such difference of such weighting factors) would be that, if the
composite signal were to be comprised only of its voice component, the
value of TX-N would almost always hover at or close to zero. The fact is,
however that the composite signal under consideration also includes its
component corresponding to the ambient noise in environment 23, and the
level of that ambient noise tends over time to remain constant or to
change only slowly. Because of the inclusion of that noise component in
the composite signal, the value of TX-N will be sustained by that
component and be determined almost entirely by the average magnitude of
the ambient noise in environment 23, only minor and relatively
insignificant fluctuations in such value being caused by the presence in
the composite signal of the voice component. It can properly be said,
therefore, that TX-N is a quantity with a value which is a relatively
significant measure of the average level of such noise (the "transmit
noise average"), while being relatively unaffected by the presence or
absence of voice sounds in such environment. The value of TX-N will
however increase in step with and accurately reflect a rise in the ambient
noise level when, as usual, such a rise occurs slowly. Moreover, the value
of TX-N will fall in step with a fall in such noise level.
In the FIG. 1 speakerphone system, the quantity TX-N is, as described,
calculated by computer 110 and stored in its memory. Then, it is on an
ongoing basis, compared in the computer with the transmit signal average
TX-S, as shown in FIG. 8, in the course of reaching decisions by the
computer of whether the system should remain in the state it is then in or
switch to another state. Such a decision calling for a change in state is
executed by the computer sending command signals to the attenuators 213,
313 to introduce into the signal paths through transmit and receive
sections 200 and 300 amounts of losses respective to these sections which
are appropriate to change the operation of the system from one state to
another. In the FIG. 1 system, the quantity TX-N is not utilized as a
factor which is determinative of the amount of loss or gain (or the amount
of change of loss or gain) introduced into the signal path in the receive
section 300.
In contrast, in the FIG. 2 speakerphone system, the quantity TX-N is
utilized as one factor among others which are determinative of the loss or
gain introduced by attenuator 313 in the path through receive section 300
for signals passing through the attenuator to speaker 70 to be reproduced
as voice sounds by it. More specifically, in the FIG. 2 system (as in the
FIG. 1 system) the computer 110 operates through its output line 360 on
programmable attenuator 313 to control the signal losses provided by it
for the purposes of switching the system from one state to another,
calibrating the system, and so on. The values of such losses are not,
however, functionally related in magnitude to the value of the transmit
noise average or other of the signals or quantities described heretofore
as involved in the normal operation of the FIG. 2 system. In the FIG. 2
system, in addition to controlling such non-magnitude related losses, the
computer 110 superposes on such losses a gain which is functionally
related in magnitude to one of such quantities, namely the transmit noise
average. Such superposition is effected as follows.
The computer derives a control signal from the quantity TX-N calculated by
the computer. Such control signal is supplied by line 360 or otherwise to
the programmable attenuator from the computer to control the attenuator
losses. The effect of such control signal on the attenuator is to
dynamically adjust, as a function of the magnitude of the quantity TX-N
(and, hence, of the average level of the audible ambient noise in
environment 23), the gain provided in the receive section for signals
passing through the attenuator to speaker 70, such gain being superposed
on the attenuator losses. The functional relationship between the
dynamically adjusted gain and such noise level is such that the gain
increases as that noise level increases and conversely. It follows from
the existence of such functional relation that, when that noise level
increases, the average level of the reproduced voice sounds also
increases. Therefore, in the face of an increase in noise level the
intelligibility of the reproduced sounds will not be lost. On the other
hand, if the average noise level in environment 23 decreases, the average
level of the reproduced sounds will correspondingly decrease in a manner
which preserves the intelligibility of the sounds but avoids their seeming
to be unduly loud.
FIG. 11 is a graphical diagram in which functional relationships between
the change in ambient average noise level in environment 23 and the
responsive dynamic change in gain effected in attenuator 313 are plotted
in vertical and horizontal coordinates representing gain and noise level,
respectively. For both coordinates, the scale used to represent changes in
the related quantity is in dB. The value of the gain as a function of the
value of the noise level is plotted as a line which may be either straight
or curved.
In FIG. 11, two gain-noise functions are plotted. The first is represented
by the straight line F.sub.1, and second by a curve F.sub.2 which
decreases in slope with increase in noise level. Both of lines F.sub.1 and
F.sub.2 are smoothed line approximations to the stepwise functional
relation between gain and noise which actually occurs in attenuator 313
because the changes in the losses (gains) provided thereby occur in 3.5 dB
steps.
In FIG. 11 with its vertical and horizontal scales in dB, the line F.sub.1
has a constant 45.degree. slope. The significance of that 45.degree. slope
is that, when the function F.sub.1 is plotted as shown in the graphical
diagram of FIG. 11 against vertical and horizontal coordinates having
scales of which the spacings between the scale numbers or other markings
are directly proportional to the actual magnitudes of the represented
quantities, then the function F.sub.1 may correspond to any of a number of
plotted lines which have various constant slopes, but which will be
straight lines representing a linear relationship between changes in the
average noise level in environment 23 and the resulting changes in the
dynamic gain of attenuator 313 produced by computer 110 in response to
those changes in noise level.
When, however, the function which is plotted in FIG. 11 is non-linear, then
that function will not be represented by a straight line when plotted in
FIG. 11. Instead as represented by the plot in that figure of the function
F.sub.2, it will, when plotted to the coordinate scales used in FIG. 11,
be in the form of a curve.
With regard to the particular gain-noise function to be used in relating
the average level of sounds reproduced by speaker 70 to the average level
of the ambient noise in environment 23, it is satisfactory to use a
functional relation of which function F.sub.1 is exemplary and the effect
of which is, as shown for function F.sub.1 to establish a linear
relationship between changes in the magnitude on an absolute scale of the
average reproduced sound level and changes in the magnitude on a absolute
scale of the average noise level. We have found, however, that as the
average noise level increases it is not necessary, in order to maintain
constant intelligibility of the reproduced sounds, for the average level
of such sound to rise in strict proportion to the noise level increase.
Rather, we have found as a surprising result that, as represented by curve
F.sub.2 in FIG. 11, constant intelligibility of the reproduced sounds can
be maintained, even as average noise level increases, if the ratio of the
increase of the average reproduced sound level to such average noise level
gradually declines with increase in such level. To state the relationship
another way, as such noise level increases, the function relating the
average level of the voice sounds reproduced by speaker 70 (or,
alternatively, the value of the dynamic gain provided by attenuator 313
for voice signals passing therethrough) is, when plotted in vertical and
horizontal ordinates having absolute magnitude scales as in FIG. 11,
desirably represented (as exemplified by line F.sub.2) by a curve which
has a slope which is positive but which decreases with increasing noise
level. In any particular use, however, the value of such slope and its
rate of decrease is implementation-dependent.
The dynamic adjustment in gain of programmable attenuator 313 will take
place in the course of each telephone call made between the FIG. 2 system
and a remote station. Accordingly, during that call, the average level of
the sounds reproduced by speaker 70 will be higher or lower in relation to
a reference value therefor when the average level of the audible ambient
noise in environment 23 is, respectively, higher or lower than a reference
value therefor. Moreover, if the average level of such noise increases or
decreases during such call, the average level of the sounds reproduced by
speaker 70 will correspondingly increase or decrease. That is, such sounds
will be characterized by the feature of automatic level control ("ALC").
What has been just been said is not true of the sounds reproduced by
speaker 75 because the voice signals from such speaker do not pass through
attenuator 313 in the course of being transferred through receive section
300 to speaker 75. Thus, the level of the sounds from the speaker 75 is
independent of the ambient noise level in environment 23. Such
independence is a desirable feature because it prevents the occurrence of
changes in the level of sounds reproduced by speaker 75 in environment 76
which (changes) have no relation to the noise level in that environment or
to any other acoustic aspect thereof.
Although the voice signals from auxiliary microphone 50 pass through
attenuator 313 in reaching speaker 70, the level of the voice sounds
reproduced by that speaker from those signals does not vary with the noise
level existing in environment 23 during the period such voice signals pass
to speaker 70. That is so because, it will be recalled, the computer 110
is on hook during such period so as not to be up-dating the quantity TX-N.
As earlier described, however, even though computer 110 is on-hook during
such period, the attenuator 313 retains therein the value of gain last set
therein in the course of the telephone call next preceding the considered
transmission of the auxiliary voice signals from microphone 50 to speaker
70. Hence, the level of the sounds reproduced from those voice signals
will have a functional relation to the level existing at the end of such
call of the ambient noise in environment 23, and will vary directly from
call to call with that end-time noise level. The functional relationship
existing in between telephone calls for auxiliary voice sounds between
their actual sound level and the noise level of the last preceding
telephone call will of course, be the same as the functional relation
existing during telephone calls between the sound level of the receive
signal from the remote station and the then current noise level in
environment 23.
FIG. 10 is a flow chart depicting steps and decisions occurring in the
process utilized in the FIG. 2 system to obtain automatic level control.
Starting consideration of the process at point 1005, as a first step 1006
the system is initialized in the sense that various conditions and
parameters of the hardware-software are established or set to prepare the
system for subsequent utilization. From step 1006 the process proceeds to
the system calibration step 1007 which has already been described.
Following system calibration, in step 1008, the programmable attenuator 313
in receive section 300 is set by computer 110 so that at time t (where t
in this step is time zero) to provide the maximum possible loss and
minimum possible gain of which attenuator 313 is capable. That minimum
gain which is set in attenuator 313 corresponds to a 65 dB value of the
level of the sounds reproduced by speaker 70. It has been found that such
65 dB level provides satisfactory sound reproduction in environments of
the reproducing speaker when the average level of the ambient noise in
that environment in 55 dB or less. That correlation between such 65 dB
sound level and such 55 dB noise level is shown at the lower ends of the
function lines F.sub.1 and F.sub.2 shown in FIG. 11.
The steps described so far precede the shown junction A and are undertaken
for the purpose of setting up the FIG. 2 system. When those steps have
been completed, the process moves on to step 1009 which is the step for
calculating on an ongoing basis the quantity or indication TX-N in the
manner previously described. Following that step is a yes-no decision 1020
undertaken with respect to TX-N (such quantity being referred to in FIG.
10 as N.sub.ambient(t)). The decision is whether or not TX-N is greater or
equal to a 55 dB reference level for noise or is less than that 55 dB
level.
If the answer is "no," process moves directly to junction C and the
speakerphone operation step 1021 so as to have the FIG. 2 system operate
in a mode which is represented by reiterated circulation around the loop
including junctions A and C and parts 1009, 1020 and 1021 of the process,
and in which mode the reproduced sound level remains at 65 dB unless and
until circulation around such loop is interrupted by the noise level
rising above 55 dB.
On the other hand, if the answer to the mentioned decision is "yes," the
process moves to junction B and then to decision 1025 where it is
determined whether or not TX-N is greater than the reference noise level
plus 31.5 dB or, in other words, 86.5 dB. It will be noted that such 31.5
dB figure corresponds to the greatest range of dB change which can be
effected by the use of the nine 3.5 dB gain or loss steps affordable by
appropriate setting of attenuator 313.
If the answer to the question relevant to decision 1025 is "yes," then the
process moves to step 1026 at which the attenuator 313 is set for minimum
loss or, to phrase it alternatively, for maximum gain.
For the circumstance, the FIG. 2 system has a mode of operation in which
the average level of the reproduced sound has reached an upper threshold
corresponding to such 86.5 dB figure and beyond which it can go no higher
despite any further increases which may occur in TX-N. The provision for
that upper threshold is necessary in order to prevent excessive gain
occurring around the loop shown in FIG. 6. The mode of operation just
mentioned can be envisaged as reiterative circulation around the loop
comprising junction A and parts 1009, 1020, 1025, 1026, and 1021 until
such time, if any, as TX-N drops below 86.5 dB.
If, on the other hand, the answer to the making of decision 1025 is "no,"
then the process moves to step 1030 at which the computer 110 as an
ongoing basis sets the attenuator 313 to have a loss equal to its maximum
loss minus an amount which is a function of the difference between TX-N
and the 55 dB reference noise level. Such setting of the attenuator is
equivalent to adjusting its gain as either a function F.sub.1 or a
function F.sub.2 (FIG. 11) of TX-N or, what is approximately the same, the
average level of the audible ambient noise sensed by the FIG. 2 system as
existing in environment 23. That is, if the FIG. 10 process arrives at
step 1030, the FIG. 2 provides the automatic level control ("ALC") which
has been described, and the operation of the system can be envisaged as
reiterative circulation around the loop comprising function A and parts
1009, 1020, 1025, 1030 and 1021 of the process.
The above-described embodiments being exemplary only, it is to be
understood that additions thereto, omissions therefrom and modifications
thereof can be made without departing from the spirit of the invention.
Thus, while the invention has been described in terms of the FIG. 2 system
which has been disclosed as a two-way communication system with
loudspeakers usable for making paging announcements in a large space such
as a factory, it will be evident that the invention (and, particularly,
the ALC feature thereof) is useful in other applications. For example, it
is noted that ALC according to the invention has a most useful application
in two-way voice communication systems in the form of cellular telephone
systems to increase the level of sounds reproduced in a vehicle as the
level of noise in such vehicle increases.
Accordingly, the invention is not to be considered as limited same as is
consonant with the recitals of the following claims.
Top